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2011-02-22Use ast_debug for console logginglathama1-5/+2
Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308526 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-21Merged revisions 308414 via svnmerge from mnicholson1-27/+21
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308416 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-21Add HTTP URI Debug logging and update noticelathama1-1/+4
enable reporting of the request URI / URL in debugging change funny debug note to a serious note. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308393 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-19Add CSS MIME Typelathama1-0/+1
Modern browsers are checking for the MIME Type of pages and in some cases will not load a file if the type is wrong. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@308330 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15No response sent for SIP CC subscribe/resubscribe request.rmudgett1-10/+29
Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307879 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Merged revisions 307535 via svnmerge from qwell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'm reverting here. (issue #18776) Reported by: alecdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307536 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback.jpeeler1-1/+5
(closes issue #18758) Reported by: rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307273 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Merged revisions 307227 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) | 11 lines Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307228 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Initialize tracking variable in structure properly. Fixes a memory leak.tilghman1-1/+3
(Reported by The_Boy_Wonder on IRC, fixed by me.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307142 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Fix issue with verbose messages not showing on remote console.qwell1-10/+8
This code was reworked recently, and since the logchannel list hadn't been created yet at this point, and it was a verbose message, it was being dropped on the floor. Now it'll continue on to where it should be handled. (closes issue #18580) Reported by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307092 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Add a couple of useful channel variables for the CC recall macro.mmichelson1-0/+3
CC_EXTEN and CC_CONTEXT will allow you to determine the channel and context that will be called when the recall occurs. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@307065 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Merged revisions 306673 via svnmerge from twilson1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@306674 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Rearrange a bit of code in the generic CC recall operation.mmichelson1-9/+11
By waiting to call the callback macro after the CC_INTERFACES, extension, priority, and context have been set, this information can be accessed more easily within the callback macro. Reported by Philippe Lindheimer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@306575 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 306123 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@306124 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 305889 via svnmerge from rmudgett2-4/+7
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305923 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-02Replacing doc/* with wiki linkslathama3-3/+3
Adding links to http(s)://wiki.asterisk.org git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305798 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Add alternative name for config option.qwell1-1/+1
The SIP sample configuration had "tlscadir" as the option name, but chan_sip used the more correct "tlscapath". Now both are accepted. Discovered (sort of) by a user on IRC in #asterisk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305247 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Asterisk HTTP response Content-typelathama1-0/+7
Address content type for BSD and other platforms (closes issue #18456) Reported by: alexo Patches: asterisk18_http.patch uploaded by alexo (license 1175) Tested by: alexo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@305083 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-31Change mutex tracking so that it only consumes memory in the core mutex ↵tilghman3-219/+279
object when it's actually being used. This reduces the overall size of a mutex which was 3016 bytes before this back down to 216 bytes (this is on 64-bit Linux with a glibc-implemented mutex). The exactness of the numbers here may vary slightly based upon how mutexes are implemented on a platform, but the long and short of it is that prior to this commit, chan_iax2 held down 98MB of memory on a 64-bit system for nothing more than a table of 32767 locks. After this commit, the same table occupies a mere 7MB of memory. (closes issue #18194) Reported by: job Patches: 20110124__issue18194.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/1066 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304950 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-28Restore some conditionals that we lost in r277814.seanbright1-5/+15
There are some cases where ast_append_ha() is called with a NULL instead of a valid int pointer. So if we get a NULL, don't try to dereference it. (closes issue #18162) Reported by: imcdona Patches: issue0018162.patch uploaded by pabelanger (license 224) Tested by: enegaard git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304638 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-27Warning message if CALLCOMPLETION(cc_callback_macro or cc_agent_dialstring) ↵rmudgett1-2/+2
are empty. Test if the value pointer is not NULL instead of not ast_strlen_zero(). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304554 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304338 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 Jan 2011) | 2 lines Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304339 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304250 via svnmerge from mmichelson1-5/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600 (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304251 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-26Merged revisions 304096 via svnmerge from seanbright1-5/+5
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines Per the man page, setvbuf() must be called before any other operation on an open file. We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304097 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-25Merged revisions 304006 via svnmerge from rmudgett1-3/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines DTMF attended transfers sometimes fail for no apparent reason. The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@304007 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-24Merged revisions 303548 via svnmerge from russell3-12/+37
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@303549 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-20Merged revision 303098 fromrmudgett1-2/+18
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines CC_INTERFACES does not get built correctly with local channels. If local channels are used with CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended to it. * Initialize the buffer to eliminate the prepended garbage. * Filter out the empty interface strings to eliminate the latter. * Added a diagnostic message if the CC_INTERFACES is ever empty. JIRA ABE-2740 JIRA SWP-2848 .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@303153 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-20Merged revisions 303106 via svnmerge from sruffell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines main/features: Use POLLPRI when waiting for events on parked channels. This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@303107 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Only check container count if it exists.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302837 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302788 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines Turn a noisy verbose message into a debug message. This can drown your console if you're using the AMI over HTTP. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302789 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Resolve a memory leak with the manager interface is disabled.russell1-1/+1
The intent of this check as it stands in previous versions of Asterisk was to check if there are any active sessions. If there were no sessions, then the function would return immediately and not bother with queueing up the manager event to be processed. Since the conversion of storing sessions in an astobj2 container, this check will always pass. I changed it to go back to checking what was intended. The side effect of this was that if the AMI is disabled, the manager event queue is populated anyway, but the code that runs to clear out the queue never runs. A producer with no consumer is a bad thing. Reported internally by kmorgan. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302785 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302693 via svnmerge from rmudgett1-12/+21
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines DTMF transfer plays the wrong sounds for wrong number or other call failure. * Set the default for features.conf.sample xferfailsound option to "beeperr" as documented instead of "pbx-invalid" and corrected the use of it in DTMF blind transfer (#1). * Improved DTMF blind transfer handling of wrong numbers. Most of the concerns in this issue were taken care of by the patch for issue 17999: Issues with DTMF triggered attended transfers. (closes issue #18379) Reported by: gincantalupo Tested by: rmudgett ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302713 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302599 via svnmerge from tilghman1-3/+58
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302634 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302554 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines Don't call strlen() when we only need to look at the next character or two. (closes issue #18042) Reported by: wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302555 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302551 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan 2011) | 7 lines Remove an extraneous \r\n at the end of a parking manager events. (closes issue #18363) Reported by: clegall_proformatique Patches: asterisk_1.8_295998_parking_manager_events_format.patch uploaded by clegall proformatique (license 1139) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302552 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-19Merged revisions 302504 via svnmerge from seanbright1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan 2011) | 7 lines Make sure that h_length is set when we short-circuit out of ast_gethostbyname. (closes issue #16135) Reported by: thedavidfactor Patches: utils.patch uploaded by thedavidfactor (license 903) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302505 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Use the expanded format type instead of plain int.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302318 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Don't enable AO2_DEBUG by default if AST_DEVMODE is on.russell1-1/+1
AO2_DEBUG is not important and is causing a false compiler warning to be generated on my Ubuntu Natty dev box. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302267 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Merged revisions 302265 via svnmerge from jpeeler1-79/+87
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip. Lock scenario presented here: Thread 1 holds ast_rdlock_contexts &conlock holds handle_statechange hints holds handle_statechange hint waiting for cb_extensionstate Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds handle_request_do &netlock holds find_call sip_pvt_ptr waiting for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 (ast_rdlock_contexts) Chan_sip has an established locking order of locking the sip_pvt and then getting the context lock. So the as stated by the summary, the operations in thread 2 have been modified to no longer require the context lock. (closes issue #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch uploaded by one47 (license 23), modified by me Review: https://reviewboard.asterisk.org/r/1072/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302266 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-18Merged revisions 302173 via svnmerge from rmudgett1-353/+661
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines Issues with DTMF triggered attended transfers. Issue #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in features.conf for attended transfer). 3) A hears MOH. B dial number C 4) C ringing. A hears MOH. 5) B hangup. A still hears MOH. C ringing. 6) A hangup. C still ringing until "atxfernoanswertimeout" expires. For v1.4 C will ring forever until C answers the dead line. (Issue #17096) Problem: When A and B hangup, C is still ringing. Issue #18395 SIP call limit of B is 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C ringing 4. Timeout waiting for C to answer 5. Recall to B fails because B has reached its call limit. Because B reached its call limit, it cannot do anything until the transfer it started completes. Issue #17273 Same scenario as issue 18395 but party B is an FXS port. Party B cannot do anything until the transfer it started completes. If B goes back off hook before C answers, B hears ringback instead of the expected dialtone. ********** Note for the issue #17273 and #18395 fix: DTMF attended transfer works within the channel bridge. Unfortunately, when either party A or B in the channel bridge hangs up, that channel is not completely hung up until the transfer completes. This is a real problem depending upon the channel technology involved. For chan_dahdi, the channel is crippled until the hangup is complete. Either the channel is not useable (analog) or the protocol disconnect messages are held up (PRI/BRI/SS7) and the media is not released. For chan_sip, a call limit of one is going to block that endpoint from any further calls until the hangup is complete. For party A this is a minor problem. The party A channel will only be in this condition while party B is dialing and when party B and C are conferring. The conversation between party B and C is expected to be a short one. Party B is either asking a question of party C or announcing party A. Also party A does not have much incentive to hangup at this point. For party B this can be a major problem during a blonde transfer. (A blonde transfer is our term for an attended transfer that is converted into a blind transfer. :)) Party B could be the operator. When party B hangs up, he assumes that he is out of the original call entirely. The party B channel will be in this condition while party C is ringing, while attempting to recall party B, and while waiting between call attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will replace the party B channel technology with a NULL channel driver to complete hanging up the party B channel technology. The consequences of this code is that the 'h' extension will not be able to access any channel technology specific information like SIP statistics for the call. ATXFER_NULL_TECH is not defined by default. ********** (closes issue #17999) Reported by: iskatel Tested by: rmudgett JIRA SWP-2246 (closes issue #17096) Reported by: gelo Tested by: rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: shihchuan Tested by: rmudgett (closes issue #17273) Reported by: grecco Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1047/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@302174 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Merged revisions 301594 via svnmerge from mnicholson1-14/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't be necessary in session_do, and removed the ms_t member from the mansession_session structure. Merged revisions 301591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines Don't store the thread id for the manager session in the structure we pass to the thread for the manager session. ABE-2543 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301595 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Merged revisions 301503 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) | 12 lines Fix CPU spike when pressing DTMF after agent login. The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301504 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Removal of unused variables so Asterisk will compile.dvossel1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301446 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Call execl() directly for a better solution for paths with spaces.tilghman1-5/+3
(closes issue #18600) Reported by: ebroad Patches: 20110111__issue18600__2.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301402 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-11Merged revisions 301307 via svnmerge from mnicholson1-11/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() ABE-2705 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301308 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-10Little endian machines were not converted properly.tilghman1-16/+16
(closes issue #18583) Reported by: jcovert Patches: 20110110__issue18583.diff.txt uploaded by tilghman (license 14) Tested by: jcovert git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@301263 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300520 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300521 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-03Merged revisions 300165 via svnmerge from rmudgett1-10/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) | 4 lines Use correct variable for atxfercallbackretries config option. * Misc formatting changes. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@300166 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-29Quote arguments, just in case there's a space in a pathname.tilghman1-1/+1
(Diagnosed by pabelanger on #asterisk-dev, fixed by me.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@299989 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-20Merged revisions 299087 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) | 5 lines Note that Park() timeout is milliseconds. (closes issue #15758) Reported by: mmurdock Tested by: mmurdock, seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@299088 f38db490-d61c-443f-a65b-d21fe96a405b
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